July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

acl list not correct?
by Ivan C Myrvold 14 Jul '08

14 Jul '08

I have the list below added at the end of acl.conf.xml, and incoming calls from 81.201.82.23, 81.201.82.25, 81.201.82.26, 81.201.82.27 gets dropped, because FreeSWITCH sends a "SIP/2.0 407 Proxy Authentication Required". But calls coming in from 81.201.82.24 and 81.201.82.28 are always successful. The range below is what Voxbone have recommended to use. Ivan <list name="voxbone" default="deny"> <node type="allow" cidr="81.201.82.0/26"/> <node type="allow" cidr="81.201.82.128/26"/> <node type="allow" cidr="81.201.83.0/26"/> <node type="allow" cidr="81.201.84.0/26"/> <node type="allow" cidr="81.201.84.128/26"/> <node type="allow" cidr="81.201.86.0/26"/> </list>

4 9

Start Up Problem
by Brian Young 14 Jul '08

14 Jul '08

When I try to start freeswitch, I get this message [root@bigblue bin]# ./freeswitch Error including /usr/local/freeswitch/conf/dialplan/extensions/*.xml Cannot Initialize [[error near line 1245]: unexpected closing tag </gateways>] I've rebuild it twice now. Any help would be greatly appreciated. Thanks

2 1

Question about IAX?
by Brian West 14 Jul '08

14 Jul '08

Just doing a quick poll to see who would like to see IAX in FreeSWITCH with all the bells and whistles? Brian West sip:brian@freeswitch.org

7 9

Trouble with SIP not found by name error
by Ryan WIlson 14 Jul '08

14 Jul '08

Hi, I'm getting the following error when I try to dial out a call to a SIP gateway. tport_by_addrinfo(0xecded60): not found by name */ 208.68.17.144:5060 I've googled the error but didn't come up with any answers. Hope someone can help! Thank you, Ryan soa_generate_offer(static::0xee2fc80, 0) called soa_static_offer_answer_action(0xee2fc80, soa_generate_offer): called soa_static(0xee2fc80, soa_generate_offer): generating local description soa_static(0xee2fc80, soa_generate_offer): upgrade with local description soa_sdp_mode_set(0x4177ba30, (nil), ""): called soa_static(0xee2fc80, soa_generate_offer): storing local description soa_get_local_sdp(static::0xee2fc80, [(nil)], [0x4177bb50], [0x4177bb4c]) called nta: selecting scheme sip sres_cache_get(0xeca0be0, NAPTR, "eico.thinktel.ca.") called rr found in cache: eico.thinktel.ca. 35 sres_cache_get(0xeca0be0, NAPTR, "eico.thinktel.ca.") returned 1 entries nta: for "eico.thinktel.ca" query "eico.thinktel.ca" NAPTR (cached) sres_cache_get(0xeca0be0, SRV, "_sip._udp.eico.thinktel.ca.") called rr found in cache: _sip._udp.eico.thinktel.ca. 33 sres_cache_get(0xeca0be0, SRV, "_sip._udp.eico.thinktel.ca.") returned 1 entries nta: for "eico.thinktel.ca" query "_sip._udp.eico.thinktel.ca" SRV (cached) sres_cache_get(0xeca0be0, SRV, "_sip._tcp.eico.thinktel.ca.") called rr found in cache: _sip._tcp.eico.thinktel.ca. 33 sres_cache_get(0xeca0be0, SRV, "_sip._tcp.eico.thinktel.ca.") returned 1 entries nta: for "eico.thinktel.ca" query "_sip._tcp.eico.thinktel.ca" SRV (cached) sres_cache_get(0xeca0be0, A, "eico.thinktel.ca.") called rr found in cache: eico.thinktel.ca. 01 sres_cache_get(0xeca0be0, A, "eico.thinktel.ca.") returned 1 entries nta: for "eico.thinktel.ca" query "eico.thinktel.ca" A (cached) nta: eico.thinktel.ca. IN A 208.68.17.144 tport_tsend(0xecded60) tpn = */208.68.17.144:5060 tport_resolve addrinfo = 208.68.17.144:5060 tport_by_addrinfo(0xecded60): not found by name */208.68.17.144:5060 tport_vsend returned 1042 nta: sent INVITE (101882567) to */208.68.17.144:5060 tport_pend(0xecded60): pending 0xee25ab0 for udp/10.0.4.5:5060 (already 0) nua(0xed0d980): call state changed: init -> calling, sent offer soa_get_local_sdp(static::0xee2fc80, [0x4177bb88], [0x4177bb80], [(nil)]) called nua: nua_application_event: entering nua(0xed0d980): sent signal r_invite nua: nua_handle_magic: entering nta: timer set next to 341 ms nta: timer A fired, retransmit INVITE (101882567) tport_tsend(0xecded60) tpn = */208.68.17.144:5060 tport_resolve addrinfo = 208.68.17.144:5060 tport_by_addrinfo(0xecded60): not found by name */208.68.17.144:5060 tport_vsend returned 1042 nta: resent INVITE (101882567) to */208.68.17.144:5060 tport_release(0xecded60): 0xee25ab0 by 0xee5e3f0 with (nil) tport_pend(0xecded60): pending 0xee25ab0 for udp/10.0.4.5:5060 (already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free nta: timer set next to 1000 ms nta: timer A fired, retransmit INVITE (101882567) tport_tsend(0xecded60) tpn = */208.68.17.144:5060 tport_resolve addrinfo = 208.68.17.144:5060 tport_by_addrinfo(0xecded60): not found by name */208.68.17.144:5060 tport_vsend returned 1042 nta: resent INVITE (101882567) to */208.68.17.144:5060 tport_release(0xecded60): 0xee25ab0 by 0xee5e3f0 with (nil) tport_pend(0xecded60): pending 0xee25ab0 for udp/10.0.4.5:5060 (already 0) nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/0 term, 0/1 free nta: timer set next to 2000 ms tport_wakeup_pri(0xecded60): events IN tport_recv_event(0xecded60) tport_recv_iovec(0xecded60) msg 0xee30a20 from (udp/10.0.4.5:5060) has 4 bytes, veclen = 1 tport_deliver(0xecded60): bad msg 0xee30a20 (4 bytes) from udp/ 10.10.0.19:5060/sip next=(nil) nta: timer A fired, retransmit INVITE (101882567) tport_tsend(0xecded60) tpn = */208.68.17.144:5060 here is the config 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <?xml version="1.0"?> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <document type="freeswitch/xml"> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <section name="configuration" description="Various Configuration"> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <configuration name='sofia.conf' description='sofia endpoint'> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <profiles> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <profile name='onepipe'> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <settings> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='nonce-ttl' value='60'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='use-rtp-timer' value='true'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='ext-sip-ip' value='64.69.80.34'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='rtp-timer-name' value='soft'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='codec-prefs' value='g729'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='debug' value='1'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='rfc2833-pt' value='101'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='ext-rtp-ip' value='64.69.80.34'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='dtmf-duration' value='100'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='sip-port' value='5060'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='dialplan' value='XML,enum'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='rtp-ip' value='10.0.4.5'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='codec-ms' value='20'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='sip-ip' value='10.0.4.5'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='accept-blind-reg' value='1'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </settings> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <gateways> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <gateway name='SBC'> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='username' value='6046309907'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='retry-seconds' value='30'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='realm' value='eico.thinktel.ca'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='extension' value='6046309907'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='caller-id-in-from' value='0'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='register' value='0'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='from-domain' value='eico.thinktel.ca'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='proxy' value='eico.thinktel.ca'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='password' value='6046309907'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] <param name='expire-seconds' value='60'/> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </gateway> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </gateways> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </profile> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </profiles> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </configuration> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </section> 2008-07-14 02:14:15+0000 [HTTPChannel,2,127.0.0.1] </document> 2008-07-14 02:14:15+ Ryan Wilson OnePipe Communications Inc. t. 778.786.2222 c. 604.716.2222 f. 866.791.7120 www.onepipe.ca

1 0

FreeSWITCH and NAT problem
by Diego Viola 14 Jul '08

14 Jul '08

Hi, I'm running FreeSWITCH on a server that has no NAT in between... the server is on Slicehost (slicehost.com) a VPS. I'm trying to register my softphone (Zoiper and X-Lite) from my computer, my computer is behind a router (NAT). So the scenario is the following: Softphone (Zoiper or X-Lite) -> router (NAT) -> INTERNET -> FS server (no nat -- public ip address). The problem is that I can't make it register, sometimes it will register and sometimes it wont, this problem is randomly. This is what I have in vars.xml: <X-PRE-PROCESS cmd="set" data="domain=maitei.com.py"/> I'm not sure if this has something to be with STUN or NAT, it works fine with Asterisk... in Asterisk I have nat=yes in my SIP user (sip.conf), is there a equivalent of nat=yes in FS that I can use for my user? Thanks, Diego

1 0

Background music in conversation
by Ashutosh 12 Jul '08

12 Jul '08

Hi, The scnario required is as follows: A calls B using FS, and a music should be played in the background while they are talking. I have searched whole wiki and fs docs and mod_dptools but unable to find a pointer so far. Even in asterisk, this has no plain solutions, other than to bring both the parties in a meetme room and starting moh for a third party on local/ channel. Anyone help me out or any links ? Best Regards, ashutosh

2 1

FreeSWITCH and Amazon EC2 port problem
by Diego Viola 12 Jul '08

12 Jul '08

Hi I have FreeSWITCH 1.0.0 installed on Amazon EC2, today was working very well... but after I re-installed FS I'm getting some problem with ports. I did followed this guide and opened the required ports: http://wiki.freeswitch.org/wiki/Amazon_ec2 The problem is when I run FS, I see that the normal ports are opened: freeswitch@Etch> sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= external profile sip:mod_sofia@67.202.9.134:5080 RUNNING (0) nat profile sip:mod_sofia@67.202.9.134:5070 RUNNING (0) default alias internal ALIASED internal profile sip:mod_sofia@10.251.203.143:5060 RUNNING (0) ec2-67-202-9-134.compute-1.amazonaws.com alias internal ALIASED outbound alias external ALIASED ================================================================================================= 3 profiles 3 aliases But when I do a netstat -a, I see that 5060 is not opened... there is a 5070 port opened though, but my softphone (X-Lite) doesn't register... what could be wrong? Thanks, Diego

2 3

freeswitch interop with Altigen PBX
by Scott Barr 12 Jul '08

12 Jul '08

Is it possible for freeswitch to communicate directly with an Altigen's PRI interface w/hardware or would freeswitch be able to send transfers as SIP INFO + flash as this is how Altigen support says the Altigen PBX handles transfer requests via SIP. Also, does anyone have any idea where I can find documentation os the specifics of using freeswitch as an SBC? Thanks. -Scott

1 0

Unvoluntary hangup, but OK after restart
by Ivan C Myrvold 11 Jul '08

11 Jul '08

My FreeSWITCH is behind nat, and I have calls coming in from internet on port 5060. This works great after I got the IP range set in acl.conf.xml. But the calls get hung up after 40-50 seconds into the call for no reason at all. But after I restart FreeSWITCH, this annoying behaviour doesn't happen. I traced both cases, when the call is hung up here: http://pastebin.freeswitch.org/4784 It looks like FreeSWITCH is resending SIP/2.0 200 OK for some reason, until the other end gives up and sends a BYE sip:Unknown@81.201.82.28 SIP/2.0 But after a restart, FreeSWITCH behaves much better, and the call goes on until one of the parties hang up normally: http://pastebin.freeswitch.org/4785 I am not sure if this is a bug, or a misconfiguration of FreeSWITCH FreeSWITCH Version 1.0.trunk (8900) Ivan

3 7

breaking the ice with IVR
by Julien Chavanton 11 Jul '08

11 Jul '08

I am making test in order to experiment IVR application with Freeswitch, I have spent a few hours trying, reading and searching the wiki. I am starting by looking for XML or Perl scripting. - How do we call a perl script from a dialplan like "default_context.xml" <extension name="perl_test"> <condition field="destination_number" expression="5000"> <action application="perl" data="/tmp/xml.pl"/> </condition> </extension> - I have tried to test the IVR application "conf/lang/en/demo/demo-ivr.xml" <extension name="ivr_demo"> <condition field="destination_number" expression="5000"> <action application="ivr" data="demo_ivr"/> </condition> </extension> error: 2008-07-11 10:04:18 [ERR] mod_dptools.c:1147 ivr_application_function() Unable to find menu Do I have to put the xml ivr file somewhere ? Thanks, Julien

2 1